[asterisk-users] new kernel and zaptel
Hi, My older kernel was 2.6.18. Now I have compiled new kernel (2.6.20). Is it necessary to re-build zaptel drivers (I'm just using ztdummy). Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new kernel and zaptel
On Mon, Mar 05, 2007 at 10:21:10AM +0200, Giedrius Augys wrote: Hi, My older kernel was 2.6.18. Now I have compiled new kernel (2.6.20). Is it necessary to re-build zaptel drivers (I'm just using ztdummy). Thanks Yes. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configurations Files of TE110P
Hello, Use the cross-over schema for creating a self cross connector. Meaning you will connect your TX pair to your RX pair. This will be the test of the physical layer of your card and the flashing red light of the led will have to turn in green. Otherwise something is not working/configured properly in your card. Best regards, ## nini @ www.modulo.ro ## younss azzayani wrote: http://www.austechpartnerships.com/forum/viewtopic.php?t=76 from the link you give me i see RJ45 pins used for E1 is 1,2,4,5 - which you may need to cross (note you can't use standard CAT5 cables here for that). what's that mean, is it mean that i have to use cable CAT6 or what exactly Can someone Give me A good schema of how to cross this cable for me 14 25 41 52 but this doesn't work ,,, :-( :-( :-(( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Problem with TE212P
Problem solved. Chris Hozian from Digium related me the problem: This problem is occurring because Asterisk expects to see the d-channel on every 16th channel. This is being offset because your TDM2400P is being loaded first. In order to fix this problem, make sure you that you are loading the kernel module for the TE212P before the TDM2400P. Then you will need to reconfigure your /etc/zaptel.conf and /etc/asterisk/zapata.conf accordingly. Your zaptel.conf should contain the following. Please keep in mind that this is only a snippet of the configuration file. -- zaptel.conf snippet: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 # the following is assuming your TDM2400P has all FXO modules fxsks=63-86 -- You will need to modify your zapata.conf to reflect these channel range changes. In addition, please verify that the jumpers on the TE212P are set for E1 mode. Now it works ok. Special thanks to Chris Hozian from Digium and Ioan Indreias from Modulo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DTMF detection problems on PRI channels?
In article [EMAIL PROTECTED], Michelle Dupuis [EMAIL PROTECTED] wrote: Sounds like the DTMF tones are too far from spec, or noisy. Is the DTMF being transcoded somewhere along the way? If you have time to killtry to separate the two frequencies in your software (I don't know goldwave) - are both present and clean and same amplitude and on freq? Remove the two frequencies and what's left? If there's a lot of noise, then the other party is doing a bad job encoding the DTMF. Otherwise we can start to chase your machine causes In the few examples I looked at recently, the audio appeared to be clean and well-formed. Notching the two frequencies left very little. The system is using E1 TDM interfaces with aLaw encoding. I see there is a define called OLD_DSP_ROUTINES which causes conditional compilation in dsp.c, and is defined or not in the Makefile. Is it worth trying the old routines? What are the main differences between the old and the new? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is the 1.0.X branch vulnerable to the SIP issue?
Thermal Wetland wrote: We are still using 1.0.7 and did not see any patches for the 1.0.X branch. Yes, it is, but we no longer provide patches (even for security issues) for the 1.0 series. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now Open
The CFP for HITBSecConf2007 - Malaysia is now open. HITBSecConf - Malaysia is the premier network security event for the region and the largest gathering of hackers in Asia. Our 2007 event is expected to attract over 700 attendees from around the world and will see 4 keynote speakers in addition to 40 deep-knowledge technical researchers presenting over two-days. Being a deep-knowledge technical conference, talks that are more technical or that discuss new and never before seen attack methods are of more interest than a subject that has been covered several times before. Summaries not exceeding 250 words should be submitted (in plain text format) to [EMAIL PROTECTED] for review and possible inclusion in the programme. Submissions are due no later than 1st May 2007. Topics of interest include, but are not limited to the following: # 3G/4G Cellular Networks # SS7/Backbone telephony networks # Analysis of network and security vulnerabilities # Firewall technologies # Intrusion detection # Data Recovery and Incident Response # GPRS and CDMA Security # Identification and Entity Authentication # Network Protocol and Analysis # Smart Card Security # Virus and Worms # WLAN and Bluetooth Security # Analysis of malicious code # Applications of cryptographic techniques # Analysis of attacks against networks and machines # File system security # Security in heterogeneous and large-scale environments PLEASE NOTE: We do not accept product or vendor related pitches. If your talk involves an advertisement for a new product or service your company is offering, please do not submit. Your submission should include: # Name, title, address, email and phone/contact number # Draft of the proposed presentation (in PDF or PowerPoint format), proof of concept for tools and exploits, etc. # Short biography, qualification, occupation, achievement and affiliations (limit 150 words). # Summary or abstract for your presentation (limit 250 words) # Time (45-60 minutes including time for discussion and questions) # Technical requirements (video, internet, wireless, audio, etc.) Each non-resident speaker will receive accommodation for 3 nights. For each non-resident speaker, HITB will cover travel expenses (through our airline partners, Malaysia Airlines) up to USD 1,000.00. HITBSecConf2007 - Malaysia: The Largest Network Security Event in Asia! http://conference.hitb.org/hitbsecconf2007kl/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now Open
The CFP for HITBSecConf2007 - Malaysia is now open. HITBSecConf - Malaysia is the premier network security event for the region and the largest gathering of hackers in Asia. Our 2007 event is expected to attract over 700 attendees from around the world and will see 4 keynote speakers in addition to 40 deep-knowledge technical researchers presenting over two-days. Being a deep-knowledge technical conference, talks that are more technical or that discuss new and never before seen attack methods are of more interest than a subject that has been covered several times before. Summaries not exceeding 250 words should be submitted (in plain text format) to [EMAIL PROTECTED] for review and possible inclusion in the programme. Submissions are due no later than 1st May 2007. Topics of interest include, but are not limited to the following: # 3G/4G Cellular Networks # SS7/Backbone telephony networks # Analysis of network and security vulnerabilities # Firewall technologies # Intrusion detection # Data Recovery and Incident Response # GPRS and CDMA Security # Identification and Entity Authentication # Network Protocol and Analysis # Smart Card Security # Virus and Worms # WLAN and Bluetooth Security # Analysis of malicious code # Applications of cryptographic techniques # Analysis of attacks against networks and machines # File system security # Security in heterogeneous and large-scale environments PLEASE NOTE: We do not accept product or vendor related pitches. If your talk involves an advertisement for a new product or service your company is offering, please do not submit. Your submission should include: # Name, title, address, email and phone/contact number # Draft of the proposed presentation (in PDF or PowerPoint format), proof of concept for tools and exploits, etc. # Short biography, qualification, occupation, achievement and affiliations (limit 150 words). # Summary or abstract for your presentation (limit 250 words) # Time (45-60 minutes including time for discussion and questions) # Technical requirements (video, internet, wireless, audio, etc.) Each non-resident speaker will receive accommodation for 3 nights. For each non-resident speaker, HITB will cover travel expenses (through our airline partners, Malaysia Airlines) up to USD 1,000.00. HITBSecConf2007 - Malaysia: The Largest Network Security Event in Asia! http://conference.hitb.org/hitbsecconf2007kl/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMS ON ASTERISK
We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5 from Counterpath). As far as we know, Asterisk don't support yet IM (Instante Message) feature,instead Eyebeam have this feature. Is that true? Is there any new version from Asterisk that supports IM? Eduardo R. Assis Soluziona Ltda Consultor Sênior - TELECOM Al. Tocantins, 125 - 290 andar - Alphaville Barueri - São Paulo-SP - CEP 06.455-020 E-mail. [EMAIL PROTECTED] Tel. +55 11-4197-0654 Fax. +55 11-4197-0660 Cel. +55 11-8577-0950 www.soluziona.com.br www.soluziona.com.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Problem with TE212P
On Mon, Mar 05, 2007 at 11:00:36AM +0100, Benito Camelas wrote: Problem solved. Chris Hozian from Digium related me the problem: This problem is occurring because Asterisk expects to see the d-channel on every 16th channel. This is being offset because your TDM2400P is being loaded first. HUH?? The dchannel needs to be no. 16 in the span, and unrelated to its general channel number, right? In order to fix this problem, make sure you that you are loading the kernel module for the TE212P before the TDM2400P. Then you will need to reconfigure your /etc/zaptel.conf and /etc/asterisk/zapata.conf accordingly. Your zaptel.conf should contain the following. Please keep in mind that this is only a snippet of the configuration file. -- zaptel.conf snippet: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 # the following is assuming your TDM2400P has all FXO modules fxsks=63-86 -- You will need to modify your zapata.conf to reflect these channel range changes. In addition, please verify that the jumpers on the TE212P are set for E1 mode. Now it works ok. Special thanks to Chris Hozian from Digium and Ioan Indreias from Modulo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CTI
I'm implementing a TAPI driver to use with CTI-TAPI application. If you are interesting vist activa.sourceforge.net 2006/11/28, Matt Florell [EMAIL PROTECTED]: Have you looked at QueueMetrics? http://queuemetrics.loway.it/ There are also several call center packages for Asterisk out there that have all of the reports built into them that you want: http://www.voip-info.org/wiki/view/Predictive+dialer MATT--- On 11/28/06, Hernany Oliveira [EMAIL PROTECTED] wrote: I need something like Call Manager. I need to know how many agents is logged in, how many calls are on queues, transfer calls, hang up calls, reports and so on. Everything related to a Call Center operation. I have been looking for and I did not find anything. -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Matt Florell Enviada em: segunda-feira, 27 de novembro de 2006 17:59 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] CTI CTI is a pretty broad term, what exactly do you need to do? Do you need to connect with another system? What features do you need? MATT--- On 11/27/06, Hernany Oliveira [EMAIL PROTECTED] wrote: Is there any cti for asterisk ?? Where may I download it ?? Thanks in advance Hernany -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.17/553 - Release Date: 27/11/2006 04:00 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.17/553 - Release Date: 27/11/2006 04:00 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.19/555 - Release Date: 27/11/2006 18:09 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A102d and Asterisk on Debian 3.1.
Hi, try to set the TDMV_DCHAN = 16 (E1) or 24 (T1). I had the same problem while updating from 2.3.4-(2|3) to 2.3.4-7 . I think, that wancfg did not set this value correctly. In my setup i updated an running system to new version, so the LinkLayer is still ok. Best Regards, Markus On 1/16/07, Klaus Darilion [EMAIL PROTECTED] wrote: Erik Forsen wrote: Did you find any solution to this problem? I have the exact same problem with a Sangoma A102d card on debian 3.1, 2.6.19 and wanpipe 2.3.4-4. I've followed several different guides, including the one on sangoma's wiki. When I try to make a call out, I get this error: Jan 16 13:17:28] WARNING[18084]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) Also got the same SABME errors as you do. Maybe you have link-layer problems - maybe you are using a wrong cable. You can test the card using a E1 cross-over cable between the 2 ports of the sangoma card. regards klaus -- Klaus Darilion nic.at ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TE212P
I agree with Tzafrir on this one. i have a digium dual span card that starts at channel nine because i have two TDM400s that load first and i have had no problems whatsoever with the D channel. On 3/5/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Mar 05, 2007 at 11:00:36AM +0100, Benito Camelas wrote: Problem solved. Chris Hozian from Digium related me the problem: This problem is occurring because Asterisk expects to see the d-channel on every 16th channel. This is being offset because your TDM2400P is being loaded first. HUH?? The dchannel needs to be no. 16 in the span, and unrelated to its general channel number, right? In order to fix this problem, make sure you that you are loading the kernel module for the TE212P before the TDM2400P. Then you will need to reconfigure your /etc/zaptel.conf and /etc/asterisk/zapata.conf accordingly. Your zaptel.conf should contain the following. Please keep in mind that this is only a snippet of the configuration file. -- zaptel.conf snippet: span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 # the following is assuming your TDM2400P has all FXO modules fxsks=63-86 -- You will need to modify your zapata.conf to reflect these channel range changes. In addition, please verify that the jumpers on the TE212P are set for E1 mode. Now it works ok. Special thanks to Chris Hozian from Digium and Ioan Indreias from Modulo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read() status?
Yes. Use show application read in the cli On 3/5/07, Yuan LIU [EMAIL PROTECTED] wrote: Does application Read() return a status? Console displays stuff, but show application read doesn't mention any status variable. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configurations Files of TE110P
Hi, the telco has given me a across cable (witch i put it on the modem so the led modem( LOS Tx became Off : that's mean that the connection is on) so i taked this cable i put it in my TE110P digium card, so the led came green but when i relayed TE110P to the modem the (led Modem turn of :mean ok) the TE110P became RED (mean not ok :-( ) the teleco sayed that the delta channel is number 16( im using just TE110P) hdb3, euroisdn so what's the problem :( i'm realy lost 2007/3/5, Ioan Indreias [EMAIL PROTECTED]: Hello, Use the cross-over schema for creating a self cross connector. Meaning you will connect your TX pair to your RX pair. This will be the test of the physical layer of your card and the flashing red light of the led will have to turn in green. Otherwise something is not working/configured properly in your card. Best regards, ## nini @ www.modulo.ro ## younss azzayani wrote: http://www.austechpartnerships.com/forum/viewtopic.php?t=76 from the link you give me i see RJ45 pins used for E1 is 1,2,4,5 - which you may need to cross (note you can't use standard CAT5 cables here for that). what's that mean, is it mean that i have to use cable CAT6 or what exactly Can someone Give me A good schema of how to cross this cable for me 14 25 41 52 but this doesn't work ,,, :-( :-( :-(( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configurations Files of TE110P
In my experience usually you want to use a straight-thru cable, not a crossover cable. Try a standard ethernet cable between TE110P and telco box. younss azzayani wrote: Hi, the telco has given me a across cable (witch i put it on the modem so the led modem( LOS Tx became Off : that's mean that the connection is on) so i taked this cable i put it in my TE110P digium card, so the led came green but when i relayed TE110P to the modem the (led Modem turn of :mean ok) the TE110P became RED (mean not ok :-( ) the teleco sayed that the delta channel is number 16( im using just TE110P) hdb3, euroisdn so what's the problem :( i'm realy lost 2007/3/5, Ioan Indreias [EMAIL PROTECTED]: Hello, Use the cross-over schema for creating a self cross connector. Meaning you will connect your TX pair to your RX pair. This will be the test of the physical layer of your card and the flashing red light of the led will have to turn in green. Otherwise something is not working/configured properly in your card. Best regards, ## nini @ www.modulo.ro ## younss azzayani wrote: http://www.austechpartnerships.com/forum/viewtopic.php?t=76 from the link you give me i see RJ45 pins used for E1 is 1,2,4,5 - which you may need to cross (note you can't use standard CAT5 cables here for that). what's that mean, is it mean that i have to use cable CAT6 or what exactly Can someone Give me A good schema of how to cross this cable for me 14 25 41 52 but this doesn't work ,,, :-( :-( :-(( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Double DTMF digits sent on IAX native bridge
Ok that makes sense, but I'm still getting double digits. It seems to me that the DTMF digit is getting detected too late. When the digit is pressed it seems like asterisk is passing the DTMF digit for a fraction of a second through the audio path and then sends the digit for however long your toneduration is set for. I can hear this happening when I dial the digits myself, I hear some kind sound being cut off for a fraction of a second and then hear the DTMF tone pass. So I guess this is why sometimes some answer machines are detecting double digits. Russell Bryant wrote: Remi Quezada wrote: I have two asterisk servers one is connected to the PSTN and the other one is connected to SIP users. The two servers connect with each other using IAX. When I have an incoming call from PSTN to the asterisk servers and have a forward to go back out to the PSTN the two IAX channel bridge together. Now every time I dial a DTMF digit, the asterisk is sending two DTMF digits. I enable debugging for iax and I do see it sending the DTMF digits two times. Here is what I see: The IAX debug that you show below only shows one of each digit. For each one, it shows Receiving the digit from one leg of the call, and then transmitting it out the other. I have spaced out your debug to separate each digit. Each one shows ... - digit - ACK - - digit --- ACK -- which is exactly what is supposed to happen. Rx-Frame Retry[ No] -- OSeqno: 018 ISeqno: 021 Type: DTMFSubclass: 1 Timestamp: 51523ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 019 Type: IAX Subclass: ACK Timestamp: 51523ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 019 ISeqno: 022 Type: DTMFSubclass: 1 Timestamp: 51543ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 020 Type: IAX Subclass: ACK Timestamp: 51543ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 019 ISeqno: 021 Type: DTMFSubclass: 2 Timestamp: 52083ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 020 Type: IAX Subclass: ACK Timestamp: 52083ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 020 ISeqno: 022 Type: DTMFSubclass: 2 Timestamp: 52103ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: IAX Subclass: ACK Timestamp: 52103ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 020 ISeqno: 021 Type: DTMFSubclass: 3 Timestamp: 52663ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 021 Type: IAX Subclass: ACK Timestamp: 52663ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 021 ISeqno: 022 Type: DTMFSubclass: 3 Timestamp: 52683ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 022 Type: IAX Subclass: ACK Timestamp: 52683ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 021 ISeqno: 021 Type: DTMFSubclass: 4 Timestamp: 53223ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 022 Type: IAX Subclass: ACK Timestamp: 53223ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 022 ISeqno: 022 Type: DTMFSubclass: 4 Timestamp: 53243ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 023 Type: IAX Subclass: ACK Timestamp: 53243ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: DTMFSubclass: 5 Timestamp: 53703ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 023 Type: IAX Subclass: ACK Timestamp: 53703ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 023 ISeqno: 022 Type: DTMFSubclass: 5 Timestamp: 53723ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 024 Type: IAX Subclass: ACK Timestamp: 53723ms SCall: 4 DCall: 16385 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 023 ISeqno: 021 Type: DTMFSubclass: 6 Timestamp: 54163ms SCall: 3 DCall: 2 [192.168.15.201:4569] Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 024 Type: IAX Subclass: ACK Timestamp: 54163ms SCall: 2 DCall: 3 [192.168.15.201:4569] Tx-Frame Retry[000] -- OSeqno: 024 ISeqno: 022 Type: DTMFSubclass: 6 Timestamp: 54183ms SCall: 16385 DCall: 4 [192.168.15.201:4569] Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 025 Type: IAX
[asterisk-users] TC400B
Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configurations Files of TE110P
i do it it doesn't work ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtsavesysname not working in 1.4
On 3/2/07, Bruce Reeves [EMAIL PROTECTED] wrote: Try renaming you column in the peers table to regserver Thanks for the suggestion Bruce, unfortunately it did not help. Any other thoughts? Does the systemname in asterisk.conf and regserver in field mysql need to be an IP address, FQDN, hostname, or what is the proper format? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configurations Files of TE110P
Hi, this is the spin config of the Teleco modem: RX - : 2 RX + : 1 Tx - : 5 Tx+: 4 what about those of TE110P what I've to do? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtsavesysname not working in 1.4
David, Here is what is working on my system, I added the following coulmn to the sip table regserver and it is varchar(20) and then set the following items in conf files. asterisk.conf systemname = server1 sip.conf displaysystemname=yes - Olle told me about this rtsavesysname=yes I bet the displaysystemname=yes is the missing setting, I seem to remeber not getting anywhere till I added that. On 3/5/07, David Thomas [EMAIL PROTECTED] wrote: On 3/2/07, Bruce Reeves [EMAIL PROTECTED] wrote: Try renaming you column in the peers table to regserver Thanks for the suggestion Bruce, unfortunately it did not help. Any other thoughts? Does the systemname in asterisk.conf and regserver in field mysql need to be an IP address, FQDN, hostname, or what is the proper format? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to disable MOH completely?
I need to disable MOH completely. We are using all SIP extensions and do not want Asterisk to invoke MOH when flash or hold is pressed on the phone. Anyone know how to configure this? Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtsavesysname not working in 1.4
Thanks again Bruce! That was indeed the problem. I added displaysystemname=yes and it started working. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P/FXS in a HP DL380 G5
The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok connector available to attach to a card that needs more power than the PCI bus can provide, like the TDM400P when FXS modules are used. HP has confirmed that there is no part they sell to give you such a connector, and Digium says their business edition folks got it to work, but only by doing nasty warranty-voiding things to the internal wiring. Has anyone figured out a solution for this? Something along the lines of an external power brick whose output attaches to a backplane slot and gives you a 12V connector inside the server? Or am I just SOL? Thanks -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Java w/ Threads
Stefan Reuter wrote: Jesus Mogollon wrote: The best option would be to use AstManProxy and connect your event manager to it. why would adding a new system in between be better than directly connecting to multiple Asterisk servers? =Stefan Simple. With the manager proxy in between, it does all the hard work of managing all the connections. It's what it's good at, and if it's doing it, I don't have to re-write all the thread management stuff again. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Java w/ Threads
Jesus Mogollon wrote: The best option would be to use AstManProxy and connect your event manager to it. I tried this. Two problems... The Asterisk Manager Proxy sends out a banner of 'Asterisk Manager Proxy/1.2' whereas the Asterisk-Java interface expects to see 'Asterisk Call Manager 1.0' (or something similar, the point is that the banner is different). So, I changed what asterisk proxy manager sends in the source. This allowed Asterisk-Java to connect... and then the Proxy manager went and core dumped . It won't work anyway. The proxy manager prefixes the name of the system to each line of output. The Asterisk-Java interface is not expecting this and will barf. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] running error: load_modules: No 'modules.conf' found - vesrion 1.4.1 from svn
Hi, I solved this issue. I run asterisk with -C parameter and it worked. So it seems that running configure with parameters changes the contents of the asterisk.conf file but doesn't change the directory path where asterisk searches for the asterisk.conf file during the start. Bests Tomasz tzieleniewski napisał(a): Hi, I have just installed the fresh svn version of asterisk and when I run it I get the following errors: [Mar 4 14:19:27] WARNING[24527]: loader.c:728 load_modules: No 'modules.conf' found, no modules will be loaded. [Mar 4 14:19:27] NOTICE[24527]: manager.c:2681 init_manager: Unable to open management configuration manager.conf. Call management disabled. [Mar 4 14:19:27] NOTICE[24527]: cdr.c:1093 do_reload: CDR simple logging enabled. I get this errors although my astetcdir contains the above considered files?? Does asterisk searched for those files in other directory than astetcdir?? the contents of my asterisk.conf: [directories] astetcdir = /home/asterisk/asterisk/asterisk astmoddir = /home/asterisk/asterisk/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk Bests Tomasz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read() status?
Yuan LIU wrote: Does application Read() return a status? Console displays stuff, but show application read doesn't mention any status variable. Yuan Liu I know that read() on a non-existent sound file will cause dial plan execution to abruptly stop (unlike background())... which is very bad imho. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 - SLA
I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. sip show subscriptions shows nothing. I just started playing with it this morning however...still playing around w/ the configs. One odd thing, I keep seeing some weirdness: [Mar 5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default (domain x.x.x.x) And also Looking for 103 Yet I have no idea where those values are coming from! I am running 1.6.7. Here is a snippet of the phone config from one of the phones: reg reg.1.displayName=Line 1 reg.1.address=station2_line1 reg.1.label=Line 1 reg.1.type=shared reg.1.thirdPartyName=2404366402-2 reg.1.auth.userId=2404366402-2 reg.1.auth.password=1234 reg.1.server.1.address= reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires=60 reg.1.server.1.register=1 I noticed that I had to set the reg.x.address field to the stationX_lineX value or the phone wouldn't fill in the icon image...but it would accept cals. Still not completely clear but I am making progress! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Friday, March 02, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 - SLA Lacy Moore - Aspendora wrote: Russell, I don't have any specifics at this time. I need to dig a little further. I'm thinking the autocontext is what is giving me fits. I can receive calls and place calls, but the hint status is not working. It currently registers as a hint showing not in use. It does not show in use. If you aren't seeing any lights change on the phones when calls are going on, check sip show subscriptions at the CLI. If the phones have not properly subscribed to the right extensions, you won't see anything. I ended up using some of the config from the bottom of the sla.txt file. The sample file may be missing the template section. The sample config does not match the config in the sla.txt. I couldn't get the sample config to work at all. Again, hopefully over the weekend I'll be able to get more information. You are correct. The sample configuration is missing the template. I will add it now. However, I just made the tarballs for 1.4.1, so this config fix didn't make it in. Using the config in the sample file, the hint status was working. I could see the line ringing, but I could not answer the lines or place calls. Using the config from the sla.txt file, I could place calls and receive calls, but the hints never showed any activity, just always not in use. As I noted earlier, check your sip show subscriptions to make sure the phones are subscribed to the right thing. Another helpful thing that you can use for debugging is to look at the output of sla show stations. You can see the state of each line appearance on each station. This should correspond with what you see on the phone ... unless there is a problem, of course. If possible, could you provide the config that you've used for testing? I'm testing using Polycom phones to try to keep things simple. I'm assuming you are using a Polycom. I have been testing with a variety of different phones. I have not tested all of the Polycom models, yet. This is one of the things we're going to have to work through. I would like to document issues with specific phones in sla.txt as we come across them. The configuration I'm using for testing looks just like the stuff in configs/sla.conf.sample. Essentially, it is: [line1] type=trunk device=Zap/3 autocontext=line1 [line2] type=trunk device=Zap/4 autocontext=line2 [station](!) type=station autocontext=sla_stations trunk=line1 trunk=line2 [station1] (station) device=SIP/station1 [station2](station) device=SIP/station2 [station3](station) device=SIP/station3 Thanks for providing some feedback on this. You are the first one to say anything about it. :) I am very eager to get everything working well so that everyone is happy. Just please be patient as I work through the initial flood of reports since it is just now getting out in the field. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P
Hi everybody, i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of TE110P and also if you can tell me have to made a cable like that?? Modem Teleco ---Self CrosscableAsterisk Rx+ -- Tx+ Rx- -- Tx- Tx+ -- Rx+ Tx- -- Rx- i'm fear that i'm gonna to damage the G723 modem card or TE110P card Thank you for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P
younss azzayani wrote: Hi everybody, i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of TE110P and also if you can tell me have to made a cable like that?? Modem Teleco ---Self CrosscableAsterisk Rx+ -- Tx+ Rx- -- Tx- Tx+ -- Rx+ Tx- -- Rx- i'm fear that i'm gonna to damage the G723 modem card or TE110P card http://www.voip-info.org/wiki/view/crossover+T1+cable http://www.cisco.com/en/US/products/hw/routers/ps233/products_tech_note09186a00800a3f09.shtml ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P
Hi everybody, i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of TE110P and also if you can tell me have to made a cable like that?? Modem Teleco ---Self CrosscableAsterisk You might check this out for a quick reference: http://www.voip-info.org/wiki/view/crossover+T1+cable -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 - SLA
Here is the debug output of the SUBSCRIBE request I am sure it has something to do with the way I am attempting to setup the Polycom for shared appearances... Nat=yes is set in the peer. I don't get these weird messages when connecting with a private line appearance. --- SIP read from x.x.x.x:60671 --- SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D From: Line 1 sip:[EMAIL PROTECTED];tag=259528A1-76B251C6 To: sip:[EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Max-Forwards: 70 Expires: 3600 Content-Length: 0 - [Mar 5 14:25:02] VERBOSE[9835] logger.c: --- (13 headers 0 lines) --- [Mar 5 14:25:02] VERBOSE[9835] logger.c: Sending to 192.168.1.116 : 5060 (no NAT) [Mar 5 14:25:02] VERBOSE[9835] logger.c: Found no matching peer or user for 'x.x.x.x:60671' [Mar 5 14:25:02] VERBOSE[9835] logger.c: Looking for 103 in default (domain x.x.x.x) [Mar 5 14:25:02] VERBOSE[9835] logger.c: --- Transmitting (no NAT) to 192.168.1.116:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D;received=x.x.x.x From: Line 1 sip:[EMAIL PROTECTED];tag=259528A1-76B251C6 To: sip:[EMAIL PROTECTED];tag=as4d77da56 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Monday, March 05, 2007 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 1.4 - SLA I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. sip show subscriptions shows nothing. I just started playing with it this morning however...still playing around w/ the configs. One odd thing, I keep seeing some weirdness: [Mar 5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default (domain x.x.x.x) And also Looking for 103 Yet I have no idea where those values are coming from! I am running 1.6.7. Here is a snippet of the phone config from one of the phones: reg reg.1.displayName=Line 1 reg.1.address=station2_line1 reg.1.label=Line 1 reg.1.type=shared reg.1.thirdPartyName=2404366402-2 reg.1.auth.userId=2404366402-2 reg.1.auth.password=1234 reg.1.server.1.address= reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires=60 reg.1.server.1.register=1 I noticed that I had to set the reg.x.address field to the stationX_lineX value or the phone wouldn't fill in the icon image...but it would accept cals. Still not completely clear but I am making progress! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Friday, March 02, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 - SLA Lacy Moore - Aspendora wrote: Russell, I don't have any specifics at this time. I need to dig a little further. I'm thinking the autocontext is what is giving me fits. I can receive calls and place calls, but the hint status is not working. It currently registers as a hint showing not in use. It does not show in use. If you aren't seeing any lights change on the phones when calls are going on, check sip show subscriptions at the CLI. If the phones have not properly subscribed to the right extensions, you won't see anything. I ended up using some of the config from the bottom of the sla.txt file. The sample file may be missing the template section. The sample config does not match the config in the sla.txt. I couldn't get the sample config to work at all. Again, hopefully over the weekend I'll be able to get more information. You are correct. The sample configuration is missing the template. I will add it now. However, I just made the tarballs for 1.4.1, so this config fix didn't make it in. Using the config in the sample file, the hint status was working. I could see the line ringing, but I could not answer the lines or place calls. Using the config from the sla.txt file, I could place calls and receive calls, but the hints never showed any activity, just always not in use. As I noted earlier, check your sip show subscriptions to make sure the phones are subscribed to the right thing. Another helpful thing that you can use for debugging is to look at the output of sla show stations.
Re: [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P
ok, here i see different config, like indian T1 crossovercable http://asterisk.pbx.in/digium-te110p-loopback-cable-india-howto i'll try this ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 - SLA
Sorry to reply to myself, once again onn the list, but since SLA is new I figured I should answer my own question before anyone else gets confused...I completely forgot about my -directory.xml defaults...so that's where all these bogus SUBSCRIBE requests were coming from. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Monday, March 05, 2007 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 1.4 - SLA Here is the debug output of the SUBSCRIBE request I am sure it has something to do with the way I am attempting to setup the Polycom for shared appearances... Nat=yes is set in the peer. I don't get these weird messages when connecting with a private line appearance. --- SIP read from x.x.x.x:60671 --- SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D From: Line 1 sip:[EMAIL PROTECTED];tag=259528A1-76B251C6 To: sip:[EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Max-Forwards: 70 Expires: 3600 Content-Length: 0 - [Mar 5 14:25:02] VERBOSE[9835] logger.c: --- (13 headers 0 lines) --- [Mar 5 14:25:02] VERBOSE[9835] logger.c: Sending to 192.168.1.116 : 5060 (no NAT) [Mar 5 14:25:02] VERBOSE[9835] logger.c: Found no matching peer or user for 'x.x.x.x:60671' [Mar 5 14:25:02] VERBOSE[9835] logger.c: Looking for 103 in default (domain x.x.x.x) [Mar 5 14:25:02] VERBOSE[9835] logger.c: --- Transmitting (no NAT) to 192.168.1.116:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D;received=x.x.x.x From: Line 1 sip:[EMAIL PROTECTED];tag=259528A1-76B251C6 To: sip:[EMAIL PROTECTED];tag=as4d77da56 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Monday, March 05, 2007 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 1.4 - SLA I have been using 2 Polycom 430s so far. I can get incoming calls just fine (both phones ring on line 1). However it doesn't appear to seize the line, so if a call is on the one phone, I can still pick up line 1 on the other and dial - and it's reflected in the connected call. I assume that's related to the hint/subscription issue Lacy indicated as well. sip show subscriptions shows nothing. I just started playing with it this morning however...still playing around w/ the configs. One odd thing, I keep seeing some weirdness: [Mar 5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default (domain x.x.x.x) And also Looking for 103 Yet I have no idea where those values are coming from! I am running 1.6.7. Here is a snippet of the phone config from one of the phones: reg reg.1.displayName=Line 1 reg.1.address=station2_line1 reg.1.label=Line 1 reg.1.type=shared reg.1.thirdPartyName=2404366402-2 reg.1.auth.userId=2404366402-2 reg.1.auth.password=1234 reg.1.server.1.address= reg.1.server.1.port= reg.1.server.1.transport=DNSnaptr reg.1.server.2.transport=DNSnaptr reg.1.server.1.expires=60 reg.1.server.1.register=1 I noticed that I had to set the reg.x.address field to the stationX_lineX value or the phone wouldn't fill in the icon image...but it would accept cals. Still not completely clear but I am making progress! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Friday, March 02, 2007 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 - SLA Lacy Moore - Aspendora wrote: Russell, I don't have any specifics at this time. I need to dig a little further. I'm thinking the autocontext is what is giving me fits. I can receive calls and place calls, but the hint status is not working. It currently registers as a hint showing not in use. It does not show in use. If you aren't seeing any lights change on the phones when calls are going on, check sip show subscriptions at the CLI. If the phones have not properly subscribed to the right extensions, you won't see anything. I ended up using some of the config from the bottom of the sla.txt file. The sample file may be missing the template section. The sample config does not match the config in the sla.txt. I couldn't get the sample config to work at all. Again, hopefully over the weekend I'll be able to get more information. You are correct. The sample configuration is missing the template. I will add it now. However, I just made the tarballs for 1.4.1, so this config fix didn't make it
[asterisk-users] Re: Asterisk Java w/ Threads
With Asterisk-Java the proposed solution to connect to multiple Asterisk servers is to create multiple AsteriskManagerConnection obeject. Each ManagerConnection handles its own thread so there is no need for custom thread handing code. All you have to do is to make sure is the EventListener objects you pass to these connections synchronize access to shared data (if there are such accesses). I think this approach is rather simple for the user and don't see a benefit in adding a proxy to that picture. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk Java w/ Threads
Stefan Reuter wrote: With Asterisk-Java the proposed solution to connect to multiple Asterisk servers is to create multiple AsteriskManagerConnection obeject. Each ManagerConnection handles its own thread so there is no need for custom thread handing code. All you have to do is to make sure is the EventListener objects you pass to these connections synchronize access to shared data (if there are such accesses). I think this approach is rather simple for the user and don't see a benefit in adding a proxy to that picture. In the past, the Asterisk Manager Interface was prone to crashes if it had more than 1 client connected to it. The proxy solved that issue. I think this issue was resolved in 1.2. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P/FXS in a HP DL380 G5
On Mon, 2007-03-05 at 12:54 -0500, James FitzGibbon wrote: The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok connector available to attach to a card that needs more power than the PCI bus can provide, like the TDM400P when FXS modules are used. HP has confirmed that there is no part they sell to give you such a connector, and Digium says their business edition folks got it to work, but only by doing nasty warranty-voiding things to the internal wiring. Has anyone figured out a solution for this? Something along the lines of an external power brick whose output attaches to a backplane slot and gives you a 12V connector inside the server? Or am I just SOL? Thanks It was exactly because of this (voiding garantee on a bunch of DL380-G4) that i had to advice to expand their Aterisk-configuration with either: 1) ordinary desktop with one or more TDM's 2) multiport ata's 3) T1 + channelbanks -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A New Phone Service - www.virtualphoneline.com
Dear Asterisk Users Mailing List - Non-Commercial Discussion, I joined VirtualPhoneLine.Com service and am really enjoying the use of it. VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, Google Talk or an IP Phone. Have a look at the http://www.virtualphoneline.com/faq and http://www.virtualphoneline.com/did for current available numbers. Follow this link http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129bOID=10 Let me know how it goes for you, Rehan Ahmed EmailID: 25 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: TDM400P/FXS in a HP DL380 G5
On 3/5/07, James FitzGibbon [EMAIL PROTECTED] wrote: Has anyone figured out a solution for this? Something along the lines of an external power brick whose output attaches to a backplane slot and gives you a 12V connector inside the server? Since i posted my original request I stumbled across this: http://www.coolerguys.com/840556029977.html They say it's custom made for them, and I certainly can't find anything else like it after several hours of searching, but it seems to be what's required. I'll have to rig up a backplate with a cutout to get the 12V connector into the case, but other than that I'm hoping it will do the trick. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable MOH completely?
Just comment everything in your musiconhold.conf On 3/5/07, David Thomas [EMAIL PROTECTED] wrote: I need to disable MOH completely. We are using all SIP extensions and do not want Asterisk to invoke MOH when flash or hold is pressed on the phone. Anyone know how to configure this? Thanks! David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk Java w/ Threads
Eric ManxPower Wieling wrote: In the past, the Asterisk Manager Interface was prone to crashes if it had more than 1 client connected to it. The proxy solved that issue. I think this issue was resolved in 1.2. Yes, this was indeed a problem with 1.0. I didn't encounter any problems regarding this for 1.2 and 1.4. Connecting from one application application to multiple Asterisk servers (which was the question) has never been a problem though. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
Follow this link http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129bOID=10 non commerical eh? care to remove that Rferreal2= part? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable MOH completely?
On 3/5/07, C F [EMAIL PROTECTED] wrote: Just comment everything in your musiconhold.conf Funny thing is, I don't have a musiconhold.conf and res_musiconhold.so is not loaded, however when I press flash or hold on my phone (connected to an ATA), on the CLI I see Asterisk try to execute music on hold. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g.729 on solaris10/x86
Hello, I'm looking for a way to have G.729 codec working on Solaris/x86. Binary codec from Digium is not compiled for Solaris/x86 (only sparc). Are there any alternative (free or commercial) G.729 implementations, which would work? I saw something from Intel and got it to compile on Linux, but it was only for evaluation purposes, so we upgraded to commercial codec from Digium. I really don't care about the U.S. patent, it does not apply here, only about copyright. If there's something with source code (could be commercial), that I can make work on Solaris, it would be great. Thank you, Juraj. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting Sip Headers From Dial App?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 This might sound strange, but is there anyway for Asterisk to set extra sip headers based on a sip phone returning a 302 in a dialplan? Example: PSTN = Asterisk = SIP-Phone, SIP-Phone returns 302 Redirect, Asterisk sets X-Something: Some_Value X-Somethingelse: Some_Other_Value, then sends the new invite with added headers. Stu Sheldon ACT USA - -- Randomly Generated Fortune Tag: Q: How do you catch a unique rabbit? A: Unique up on it! Q: How do you catch a tame rabbit? A: The tame way! -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQFF7KIhK69Y+xPZrWYRAgmWAJ9FvL+BqRr5YzXSYlkn9vLu4mHq2ACfaKrc LJts0IptsnzfawJzMNWibnM= =Kjdf -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Instant Messaging with SIP Softphone Eyebeam (was: SMS ON ASTERISK)
Am Montag, den 05.03.2007, 09:01 -0300 schrieb Assis, Eduardo: We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5 from Counterpath). As far as we know, Asterisk don't support yet IM (Instante Message) feature,instead Eyebeam have this feature. I cross-read their handbook and did not find anything about the Instant-Messaging feature besides its existence and how to send a message. I _guess_ it works with SIP messages (like SendText and ReceiveText Asterisk commands, perhaps?), but it surely is NOT related to the SMS application of Asterisk. app_sms supports landline SMS over analogue and ISDN lines, meaning the short message is encoded in a few-second 1200bps modem conversation. Of course there are more appropriate ways to send messages if you are on an IP network :-) If I were you, I would try to fiddle with the SendText and ReceiveText stuff (the latter might be available in AGI only - there are docs on the voip-info.org website, and google is our friend). SendText might only work while in a call, like exten = 1,1,Answer exten = 1,2,SendText(Lorem ipsum dolor) (send text to the caller's display) I cannot guarantee that it will work at all, there might still be other means off messaging that I just do not have an idea of. Did you test at all wether messages go anywhere, or what happens if you send a message? Best regards, Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
Or the fact that www.virtualphoneline.com is part of DIDXchange and of course you love it since you work for supertec.com, didxchange.com, and virtualphoneline.com On 3/5/07, Singer Wang [EMAIL PROTECTED] wrote: Follow this link http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129bOID=10 non commerical eh? care to remove that Rferreal2= part? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail question
Group In voicemail.conf I would like to having the following setup per context not per-mailbox settings serveremail userscontext fromstring usedirectory emailbody pagerfromstring dialout sendvoicemail callback review operator volgain nextaftercmd forcename forcegreetings tempgreetwarn Can this be done? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
Or that its not even a new service? On 3/5/07, Bruce Reeves [EMAIL PROTECTED] wrote: Or the fact that www.virtualphoneline.com is part of DIDXchange and of course you love it since you work for supertec.com, didxchange.com, and virtualphoneline.com On 3/5/07, Singer Wang [EMAIL PROTECTED] wrote: Follow this link http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129bOID=10 non commerical eh? care to remove that Rferreal2= part? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisk as Voicemail Server on a dinosaur Meridian System
We have a dinosaur Meridian system (~version 2) with 4 digital lines going to a Repartee Voicemail server. The Repartee got smoked by lightning two days ago and I'm itching to get Asterisk installed in its place. PRI is not an option since the system is so old that it doesn't even support PRI. I need to figure out how to connect the old Meridian to Asterisk otherwise. Any advice on getting Asterisk to work in its place is really appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Minor bug-fix release, no new functionality. Bugs fixed: * app_cbmysql would fail to load * Incorrect handling of recurring conferences that spanned a DST transition Minor cleanup: * A couple image files were duplicated with both upper and lowercase names. The uppercase variants were deleted and the HTML code cleaned up to use just the remaining files. The new release can be found at: http://sourceforge.net/projects/web-meetme/ We do have a volunteer developer who will be maintaining the 2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and features that are not Asterisk version dependant will still be made available for older installations. The 2.X.X chain does not have the problem with app_cbmysql, but may suffer from the DST transition bug. Thanks, The Web-MeetMe development team... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
On Tue, 06 Mar 2007 05:12:03, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Dear Asterisk Users Mailing List - Non-Commercial Discussion, I joined VirtualPhoneLine.Com service and am really enjoying the use of it. VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, Google Talk or an IP Phone. Have a look at the http://www.virtualphoneline.com/faq and http://www.virtualphoneline.com/did for current available numbers. Follow this link http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129bOID=10 Let me know how it goes for you, Rehan Ahmed Come on Rehan... Do you think we're really going to fall for that trick. We all know you represent virtualphoneline.com. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???
Hello All , I'd usually just take the latest timestamped tarballs use them , But this has gotten me a tad setback . I want to build astersik-1.4.1 I am not sure which of these is going to work correctly . Anyone else have a better idea than me ? Rsvp , Tia , JimL -rw-r--r--1 00 9397296 Feb 21 01:07 asterisk-core-sounds-es-g722-1.4.6.tar.gz -rw-r--r--1 00 2129399 Feb 21 01:07 asterisk-core-sounds-es-gsm-1.4.6.tar.gz -rw-r--r--1 0012219353 Feb 21 01:07 asterisk-core-sounds-en-alaw-1.4.6.tar.gz -rw-r--r--1 00 7167005 Feb 21 01:07 asterisk-core-sounds-fr-g722-1.4.6.tar.gz -rw-r--r--1 00 6874032 Feb 21 01:07 asterisk-core-sounds-en-g729-1.4.6.tar.gz -rw-r--r--1 00 1623436 Feb 21 01:07 asterisk-core-sounds-fr-gsm-1.4.6.tar.gz -rw-r--r--1 0018603291 Feb 21 01:07 asterisk-core-sounds-es-wav-1.4.6.tar.gz -rw-r--r--1 0012278506 Feb 21 01:07 asterisk-core-sounds-en-ulaw-1.4.6.tar.gz drwxr-xr-x3 008192 Feb 22 00:32 . -rw-r--r--1 0027839721 Feb 22 00:32 asterisk-extra-sounds-en-wav-1.4.5.tar.gz -rw-r--r--1 001375 Feb 22 00:32 asterisk-extra-sounds-en-ulaw-1.4.5.tar.gz -rw-r--r--1 0013675929 Feb 22 00:32 asterisk-extra-sounds-en-g722-1.4.5.tar.gz -rw-r--r--1 00 3235653 Feb 22 00:32 asterisk-extra-sounds-en-gsm-1.4.5.tar.gz -rw-r--r--1 0013473844 Feb 22 00:32 asterisk-extra-sounds-en-alaw-1.4.5.tar.gz -rw-r--r--1 00 2017747 Feb 22 00:32 asterisk-extra-sounds-en-g729-1.4.5.tar.gz drwxr-xr-x4 004096 Feb 22 00:40 .. drwxr-xr-x6 004096 Mar 06 00:50 .svn ncftp ...ephony/sounds/releases dir -alrt -- +-+ | James W. Laferriere | System Techniques | Give me VMS | | NetworkEngineer | 663 Beaumont Blvd | Give me Linux | | [EMAIL PROTECTED] | Pacifica, CA. 94044 | only on AXP | +-+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable MOH completely?
Could be its trying but does it actualy play the music? On 3/5/07, David Thomas [EMAIL PROTECTED] wrote: On 3/5/07, C F [EMAIL PROTECTED] wrote: Just comment everything in your musiconhold.conf Funny thing is, I don't have a musiconhold.conf and res_musiconhold.so is not loaded, however when I press flash or hold on my phone (connected to an ATA), on the CLI I see Asterisk try to execute music on hold. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
On Tue, 06 Mar 2007 05:12:03, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Dear Asterisk Users Mailing List - Non-Commercial Discussion, I joined VirtualPhoneLine.Com service and am really enjoying the use of it. VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, Google Talk or an IP Phone. Have a look at the http://www.virtualphoneline.com/faq and http://www.virtualphoneline.com/did for current available numbers. Follow this link http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129bOID=10 Let me know how it goes for you, Rehan Ahmed Come on Rehan... Do you think we're really going to fall for that trick. We all know you represent virtualphoneline.com. Regards, David Now Linda has to go on a new PR campaign.. This is what happens when out source for cheaper rates ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com
Rehan Ahmed Come on Rehan... Do you think we're really going to fall for that trick. We all know you represent virtualphoneline.com. he is so clueless I can't believe his companies are still in business Regards, David ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Questions
Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address=10.0.17.8 dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan from phone1.cfg: dialplan dialplan.1.impossibleMatchHandling=0 dialplan.1.removeEndOfDial=1 dialplan.2.impossibleMatchHandling=0 dialplan.2.removeEndOfDial=1 dialplan.3.impossibleMatchHandling=0 dialplan.3.removeEndOfDial=1 dialplan.4.impossibleMatchHandling=0 dialplan.4.removeEndOfDial=1 dialplan.5.impossibleMatchHandling=0 dialplan.5.removeEndOfDial=1 dialplan.6.impossibleMatchHandling=0 dialplan.6.removeEndOfDial=1 digitmap dialplan.1.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx dialplan.1.digitmap.timeOut=3 dialplan.2.digitmap= dialplan.2.digitmap.timeOut= dialplan.3.digitmap= dialplan.3.digitmap.timeOut= dialplan.4.digitmap= dialplan.4.digitmap.timeOut= dialplan.5.digitmap= dialplan.5.digitmap.timeOut= dialplan.6.digitmap= dialplan.6.digitmap.timeOut=/ routing server dialplan.1.routing.server.1.address=10.0.17.8 dialplan.1.routing.server.1.port=5060 dialplan.2.routing.server.1.address= dialplan.2.routing.server.1.port= dialplan.3.routing.server.1.address= dialplan.3.routing.server.1.port= dialplan.4.routing.server.1.address= dialplan.4.routing.server.1.port= dialplan.5.routing.server.1.address= dialplan.5.routing.server.1.port= dialplan.6.routing.server.1.address= dialplan.6.routing.server.1.port=/ emergency dialplan.1.routing.emergency.1.value= dialplan.1.routing.emergency.1.server.1= dialplan.2.routing.emergency.1.value= dialplan.2.routing.emergency.1.server.1= dialplan.3.routing.emergency.1.value= dialplan.3.routing.emergency.1.server.1= dialplan.4.routing.emergency.1.value= dialplan.4.routing.emergency.1.server.1= dialplan.5.routing.emergency.1.value= dialplan.5.routing.emergency.1.server.1= dialplan.6.routing.emergency.1.value= dialplan.6.routing.emergency.1.server.1=/ /routing /dialplan Thanks in advance! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable MOH completely?
On 3/5/07, C F [EMAIL PROTECTED] wrote: Could be its trying but does it actualy play the music? It's not actually playing anything. I guess it just seems odd that Asterisk re-invites the media back to itself when a call is put on hold (when MOH is disabled), instead of simply disconnecting the media until the call is retrieved. I guess I was hoping for a config option that would simply turn MOH off to achieve this behavior. Does such a config option exist in 1.4? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] server generated outbound conference calls?
Is anyone currently generating asterisk server outbound conference calls via some form of desktop application or IM client? What I mean by this is can I currently initiate an event on my asterisk server where it dials me first as a conference initiator and then 4 of my contacts by me either; 1. Right clicking in Outlook on their names and highlighting join conference command? or 2. Dropping and dragging multiple Outlook icons onto a desktop application (such as hud-light) or 3. Using some form of IM (either Windows Messenger, Jive, Gtalk or Jabber client) initiate conference calls between multiple buddies? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2, DTMF and x86_64.
Hi all, I'm just starting to play with 1.4. I installed 1.4.1 on an Ia32 machine, and can't find any problems. So, I decided to upgrade my home pbx. All went well until I tried using my S101 to talk to the IVR. Some times, the first one or two digits get through, but eventually a digit will get stuck, playing continuously until the call is terminated. I have confirmed this on another x86_64 machine that I connect with. Also, when I reloaded IAX2, Asterisk crashed with a message about a double linklist and an ugly trace. Unfortunately, the crash didn't make it into the logs. Any ideas? -- Bill in Denver ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_queue not using exit context?
Before I report this as a bug (and get whacked with more bad karma), I'd
like to make sure I'm understanding this feature.
I'm defining a queue with a couple of SIP phones as the memebers -- not
agents.
queue.conf allows you to set an exit context such that if set (and you use
the T or t option) allows the caller or callee to transfer the call to
that context/extension.
This doesn't appear to be working for me.
If I set T, app_queue looks for the extension in the caller's current
context -- where the queue() call is.
If I set t, app_queue looks for the extension in the callee's context --
in my case, the context specified in sip.conf.
Am I misunderstanding how this is supposed to work?
Is it working for you?
Relevant snippet from my queue.conf:
[customer-service]
context = customer-service
member = sip/pap2-000F66A83C90-line-1
Relevant snippet from my sip.conf:
[pap2-000F66A83C90-line-1]
context = inside
type= friend
Relevant snippet from my extensions.conf:
[first-time-caller](h,s)
exten = s,n,
set(CALLERID(number)=${CARD-NUMBER:0:4}${CARD-NUMBER:-4})
exten = s,n,queue(customer-service|nrt)
[customer-service](h,i,s)
exten = s,n,hangup
exten = 3,1,goto(redirect,s,1)
exten = 7,1,goto(theme,s,1)
exten = 8,1,goto(enter-card-number,s,1)
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
___
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[asterisk-users] Re: Registrations, how many is too many?
voiplist wrote: We do not use dyndns for anything, not sure what we would even use it for. We do have lots of hostnames to different systems in our sip.conf, I have changed them all to IP to see if this helps. So, you think that maybe when DNS gets hosed up that it could cause SIP to just tank on a high volume system? Of course you need to have DNS server installed on Asterisk machine. So, Asterisk will ask that machine for DNS records. If that machine doesn't know the answer (because Internet connection is down), at least Asterisk will get fast answer so it won't stop responding. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 1.4 lost internet internal phones loose registration
Thomas Kenyon wrote: Asterisk also seems to barf if it makes a registration/renewal request and it doesn't receive a reply in a timely fashion which will obviously happen if your internet connection disappears. (all versions I've used). That's why people should use dnsmasq. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED
Kristian Kielhofner wrote: Hey everyone, I came across a situation where I needed to use CDP to advertise a voice vlan to Polycom/Cisco (and other CDP capable phones) without a Cisco switch. Hi Kristian! Thank you for your work. I'm not able to test this right now, but I'll sourly need this sometimes. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
